similar to: random digits dialing during call

Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

Hi List, Has anyone heard of an ATA device that supports TCP & TLS? Not much comes up in searching, thought to check here for some device suggestions. TIA, Skyler

Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. S. -------------- next part -------------- An HTML

Hello. I am trying to write an interactive function that asks the user for a vector of observations. Unfortunately, if a user inputs a vector, R treats the vector name as a string instead of a variable. Here is an example: vector.input<-function(){ k<-as.integer(readline("Input number of vectors: ")) obs<-as.integer(readline("Input number of observations per vector

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Hi, I was wondering if there is already an application or a simple mechanism to convert the dialed extension into digits if letters were used. I don't know if there is a name for that, I mean the letters on the phone keypad: ABC=2, DEF=3, ... So when I call e.g. JOE, the extension 563 shall be used. Do I need to write my own little application to accomplish this? Thanks, Armin

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have

System config:- bash-3.2# ifconfig -a lo0: flags=2001000849<UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL> mtu 8232 index 1 inet 127.0.0.1 netmask ff000000 rge0: flags=201004843<UP,BROADCAST,RUNNING,MULTICAST,DHCP,IPv4,CoS> mtu 1500 index 2 inet 192.168.1.53 netmask ffffff00 broadcast 192.168.1.255 ether 0:1e:8c:25:cc:a5 lo0:

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

Hi, I''m a bit unsure as to when one uses a nested resource and when one uses a normal resource. If you have a belongs_to, has_one/many relationship between models is that automatically an indication of a nested resource or can these resources still be represented in the normal resource way? I have a resource (talker) that belongs_to a number of other models (network, data_date,